asterisk users - Nov 2008 - directrtpsetup without reinvite

Hi,

    I want to be able to bridge two sip channels using direct RTP 
between my endpoints (Audio IP : not local) but without
using reinvites. So I set up my asterisk sip endpoints as follows:

[test1]
type=friend
host=dynamic
username=test1
dtmfmode=info
context=test_rtp
allow=all
canreinvite=no
directrtpsetup=yes

[test2]
type=friend
host=dynamic
username=test2
dtmfmode=info
context=test_rtp
allow=all
canreinvite=no
directrtpsetup=yes

... but it doesn't work. How can I ensure that the RTP is not going 
through my asterisk box and that the re-invite method is not used?

P.S. Both endpoints are using the same codec, so no codec translation 
takes place.

On Mon, Nov 10, 2008 at 8:13 AM, regs at kinetix.gr <regs at kinetix.gr>
wrote:
> Hi,
>
>    I want to be able to bridge two sip channels using direct RTP
> between my endpoints (Audio IP : not local) but without
> using reinvites. So I set up my asterisk sip endpoints as follows:
>
> [test1]
> type=friend
> host=dynamic
> username=test1
> dtmfmode=info
> context=test_rtp
> allow=all
> canreinvite=no
> directrtpsetup=yes
>
> [test2]
> type=friend
> host=dynamic
> username=test2
> dtmfmode=info
> context=test_rtp
> allow=all
> canreinvite=no
> directrtpsetup=yes
>
> ... but it doesn't work. How can I ensure that the RTP is not going
> through my asterisk box and that the re-invite method is not used?
>
> P.S. Both endpoints are using the same codec, so no codec translation
> takes place.
>
What version of Asterisk is this?  Last I heard (from Olle) this
option was very experimental and should not be used on production
systems.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

Kristian Kielhofner wrote:
> What version of Asterisk is this?  Last I heard (from Olle) this
> option was very experimental and should not be used on production
> systems.
Oh.

Well.

That throws a wrench into the gears of a few uses of Asterisk as a 
dedicated signaling-only B2BUA I was planning on.

-- 
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

Kristian Kielhofner wrote:
> What version of Asterisk is this?  Last I heard (from Olle) this
> option was very experimental and should not be used on production
> systems.
He even helpfully documented it that way in the sip.conf.sample file,
along with a list of (known) cases where it will fail, although there
are probably plenty more.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)